From 8706e767affc6bdc8cb7a67ca3a20a62779ff048 Mon Sep 17 00:00:00 2001 From: 雾聪 <wucong.lyb@alibaba-inc.com> Date: 星期三, 17 五月 2023 15:45:46 +0800 Subject: [PATCH] Merge branch 'main' of https://github.com/alibaba-damo-academy/FunASR into main --- funasr/runtime/python/websocket/ws_server_online.py | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/funasr/runtime/python/websocket/ws_server_online.py b/funasr/runtime/python/websocket/ws_server_online.py index ba50f62..16a3abe 100644 --- a/funasr/runtime/python/websocket/ws_server_online.py +++ b/funasr/runtime/python/websocket/ws_server_online.py @@ -53,6 +53,9 @@ if "is_speaking" in messagejson: websocket.is_speaking = messagejson["is_speaking"] websocket.param_dict_asr_online["is_final"] = not websocket.is_speaking + # need to fire engine manually if no data received any more + if not websocket.is_speaking: + await async_asr_online(websocket,b"") if "chunk_interval" in messagejson: websocket.chunk_interval=messagejson["chunk_interval"] if "wav_name" in messagejson: @@ -82,7 +85,7 @@ async def async_asr_online(websocket,audio_in): - if len(audio_in) > 0: + if len(audio_in) >=0: audio_in = load_bytes(audio_in) rec_result = inference_pipeline_asr_online(audio_in=audio_in, param_dict=websocket.param_dict_asr_online) -- Gitblit v1.9.1